Hi everyone,
Newbie here. PBXIAF 2.90.2 with all the updates (installed 2 weeks ago). Using voip.ms for testing on a sip trunk. Server is dual honed behind a firewall. Phone network is different than lan for the server. Everything is working ok. Got couple of sip phones and soft phones. When I try to dial out i get the following voice message:
that number has not yet been assigned please contact technical support
After several retries though, the call goes through. It does it from all phones including the soft phones. No dial rules on the trunk. Two single out bound routes, one for international (just one entry 011.) and one for all others.
I copied and pasted the setup suggested by the voip.ms wiki
I don’t see anything that stands out in the logs either.
It looks like I needed to put in dialing rules on my sip trunk. I thought that if we had no rules then it would simply pass the number as is to the sip provider, is this not true then? I was trying to keep this as simple as possible for my users. On a regular bell land line we can dial 305, 786, 954, and 561 without needing a 1. Since I am using voip.ms what rules should I have put in. So far I have these:
305NXXXXXX
786NXXXXXX
954NXXXXXX
561NXXXXXX
(1)786NXXXXXX
I found this thread:
http://pbxinaflash.com/forum/showthread.php?t=7324
I cleared all of my dialing rules on the voipms trunk and left just
(1)+NXXNXXXXXX
I think that may have worked. I will up date this as I go.
Two issues that I had:
One issue was my dialing rules were not correct: I deleted all of the dialing plans for the sip trunk, setup a simple dialing pattern in my outbound routes.
The second issue had to do with my sip trunk setting, I kept getting all circuits are busy: Contacted Voipms support and reset my trunk settings to basics. Changed from tampa.voip.ms to the ip address and it is working.